[asterisk-users] feedback mechanism
Hi All, I would like to write a script to run on peers to monitor my resources such as whether a card was removed and send a signal to LB so it can resize the capacity configuration for that peer, but I have no idea which event in Asterisk should be monitored when a card was remove or added? Someone here pointed me out to AMI, but which one? and how can I send the signal to LB? so if you have any idea, site or resources to point me out, I would gladly appreciate. Thanks. -- Lito A. Lampitoc http://www.godlessgeek.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] feedback mechanism
Hello guys, Is Asterisk capable of sending feedback to a load balancer, such as, notifying LB when maximum capacity of Asterisk server has change (like a GW with more or less E1 cards)? -- Lito A. Lampitoc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] system recording problem using wav file
When I upload a pre recorded wav file using trixbox, it can't be played on the welcome message. But when I record using xlite, it works ok. trixbox required 8Khz PCM 16bit recording, I used it, but still no success. Any idea? Thanks. Lito ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: how to define a pilot number
thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? On 3/27/07, David Cook [EMAIL PROTECTED] wrote: is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Lito I'm assuming you are talking about analog lines as PRI's will do this more-or-less naturally. This is a telco feature as opposed to an Asterisk feature. Here in Bell Canada country they call it Ringer Equivalence. Call your local carrier and they should be able to tell you what they call it in their marketing world. You tell the telco which lines you want the calls to roll to then all three will terminate calls to the pilot number. Now it doesn't work exactly as you had described - it doesn't move the call so as to free up the first port. It merely says the first port is busy and terminates the next call on the next port in sequence. This means you can't count on which line is available at any time. For outbound, you need to put the three lines in an Asterisk group and test the group for availability to select an available line to dial out on. dbc. -- David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to define a pilot number
Hello all, is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What card for E1R2?
Hi all, My client has an E1r2 connection (10 channels), what Digium card do I need? Thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What card for E1R2?
Sorry, i mean 30 channels. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What card for E1R2?
Yes. I will take a look at it. Thanks for the suggestion Lito On 11/20/06, Josué Conti [EMAIL PROTECTED] wrote: Hi Lito, as good? If need to necessary a Digium Wild Card TE110P and the libraries to protocol mfc/r2. I like libraries developed for Steve Underwood, where it places for download in the site www.soft-switch.org . I hope this help Regards Josué 2006/11/19, Lito Lampitoc [EMAIL PROTECTED]: Sorry, i mean 30 channels. ___ --Bandwidth and Colocation provided by Easynews.comhttp://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tdm2400p question
Hi all,I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines.6 plus 6 is 12, how come it's 24?if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate.thanks.Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm2400p question
I see, thank you very much for all your answers. Btw, the interface looks different than the ordinary rj45, so how are you going to plug in the rj45 plug to it?On 10/16/06, George Pajari [EMAIL PROTECTED] wrote: The TDM2400P supports up to six quad modules -- each quad modulesupports EITHER four FXS ports OR four FXO ports...THEREFOREwith 6 quad FXO modules one has 24 FXO ports,with 5 quad FXO modules and 1 quad FXS module one has 20 FXO ports and 4 FXS ports...the remainder of these examples is left as an exercise for the reader.The board does not have to be fully populated (i.e. you do not need tohave all six quad module positions filled). g.--George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)Open Source VoIP/Telephony Specialists1 877 NET VOIP (638 8647 x102) www.netvoice.cawww.ip-centrex.cawww.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and hipath 3750
Hi All,I need to connect my hipath 3750 to asterisk. I have 4 hg1500 in my hipath already. I read that it is possible to do it via h.323. Asterisk must be in front of hipath 3750. Asterisk === hg1500 hipath 3750. other people in the list says that TMS2 can also be used.Can anybody refer me to any how-to or any reference to a successful implentation? Any help is highly appreciated.Thank you very much. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] recommended hardware specs
I am connecting 80 locals with 16 PSTN lines. Which means, i need 4 digium cards with 4 FXOs per card. All 80 locals will be connected to ATA devices.What do you guys recommend for the hardware specs for this kind of setup? Thanks.Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cannot received calls in pstn line
sorry for my english, but here' s the scenario:I have a 1 FXO and 1 FXS. when my telephone (direct line) is connected to the FXO, I cannot receive an incoming call. Since I am in an office with conventional PBX, I tried to connect one local line (local to PBX) to the FXO and made a call from other direct lines (outside the office) and it works! brandon, i'll try your suggestion.thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cannot received calls in pstn line
Hi All, I'm having problems receiving calls in my direct lines, but it's working fine in local lines (extensions). When a direct line is connected to my fxo it can't handle the call, but when an extension is connected it's ok. Any suggestion? thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call forwarding to mobile phone
Hello all,Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated.thanksLito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding to mobile phone
is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.On 7/18/06, Sam Tam [EMAIL PROTECTED] wrote: Get an GSM Gateway from cyber-telecom.net From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Lito Lampitoc Sent: Tuesday, July 18, 2006 4:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] call forwarding to mobile phone Hello all, Is it possible to forward a call received by the asterisk server to a mobile phone? If yes, how? a link or reference is highly appreciated. thanks Lito ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call forwarding to mobile phone
thanks a lot!On 7/19/06, Woodoo People .pGa! [EMAIL PROTECTED] wrote: is there a way I can do call forwarding to mobile phone without using a gsm gateway? my landline is capable of calling a gsm network.[from-gsm]exten = s,1,Dial(Zap/$your_mobile)that's all --WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com[EMAIL PROTECTED]]iCQ#33118021[wpeople.on.iRCNet [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: siemens pbx and asterisk
Do I still need an ATA adapter for my analog phones once I was able to connect my Siemens HiPath 3750 to Asterisk?Thanks in advance.On 6/27/06, richard Coco [EMAIL PROTECTED] wrote: hi all,The HG3550 V1 and HG3550v1.1 only supports H.323 V.2.I'am not sure but i thing that the feature CallerIDName was introduced in version 3 of the H.323standard. More informations about the owerviews at http://www.packetizer.com/voip/h323/.-Concerning HiPathv3.0.In version 3.0 the HiPath has a new board (the HG3540)which supports SIP (for Endpoints) and SIPQ for SIP-trunking. You are now able to interconnectAsterisk and HiPath using H.323, ISDN and/or SIPQ.rich--- Herchi Silviu [EMAIL PROTECTED] wrote: Hi, As I wrote, the HiPath needs to be upgraded to version 3 (don't ask me any details, I'm not a Siemens expert) in order to have the CallerID name passed over the H.323 link. Earlier versions (my case) ony sends and accepts the CallerId number. I have set up a workaround for calls coming to Asterisk: an AGI script sets the CallerID name according to their CallerID number by looking it up in a database. This is done in real time for every incoming call. Obviously it doesn't work for calls going from Asterisk to the HiPath. Regards, Silviu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Michael Hamann Sent: 27 June 2006 14:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: siemens pbx and asterisk Hi Silviu, did you manage to get the callername to work? I have a comparable setup with a hipath System but I can�t get the callername to be displayed over the trunk. The callernumber works but not the name... Any suggestion? Thanks Michael We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers. On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature - the callerid name is not transmitted over the link (it is a limitation of the PBX that should disappear when it is upgraded to the V3 version). The nice thing is it doesn't take any special hardware on the Asterisk server - you just have to compile and setup an H.323 channel (asterisk-oh323 works best for us). On the second one we have a Digium TE110P connected to the PBX using a PRI. It works well too, you just need the PBX to have a trunk defined and you're ready to go. We only use ten channels, so I can't say if the performance is better. In this case you need libpri and zaptel on the Asterisk. I hope this helps, Silviu --- Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users__Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection around http://mail.yahoo.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hipath 3750
Hello all,My Siemens PBX is hipath 3750, since HG3550 i think is applicable only to hipath 4000 for interfacing with asterisk,what do you think will I needing for asterisk and hipath 3750?Thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hipath 3750 + hg1500 + asterisk
Has anyone here successfully tried this?hipath 3750 -- hg1500 -- asteriski'm not sure with the flowlines though.Thanks.Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] siemens pbx and asterisk
Hello all,I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks.Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: siemens pbx and asterisk
Hello Silviu,Thank you very much for your reply. I will try that.On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi Lito, We have successfully integrated an existing Siemens HiPath 4500 PBX with two Asterisk servers. On the first one we use a H.323 trunk (it needs a card on the PBX, I think it's called HG3550). It works pretty well, except for one missing feature - the callerid name is not transmitted over the link (it is a limitation of the PBX that should disappear when it is upgraded to the V3 version). The nice thing is it doesn't take any special hardware on the Asterisk server - you just have to compile and setup an H.323 channel (asterisk-oh323 works best for us). On the second one we have a Digium TE110P connected to the PBX using a PRI. It works well too, you just need the PBX to have a trunk defined and you're ready to go. We only use ten channels, so I can't say if the performance is better. In this case you need libpri and zaptel on the Asterisk. I hope this helps, Silviu --- Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk to mobile phone
Is it possible to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 mobile phones brought by the engineers in the field, when someone calls the trunkline, asterisk will hunt which of the 10 mobile phones is available. What do I need for this setup? Thanks in advance.Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to mobile phone
what brand of gsm gateway do you think works well with asterisk?On 6/27/06, Colin Anderson [EMAIL PROTECTED] wrote: A GSM gateway will allow you to specify a ruleset so a channel on the gateway is always locked to a particular mobile number, then you just send the call from Asterisk to the gateway and it will do the hunt for you. -Original Message-From: Lito Lampitoc [mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 27, 2006 7:59 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] asterisk to mobile phoneIs it possible to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 mobile phones brought by the engineers in the field, when someone calls the trunkline, asterisk will hunt which of the 10 mobile phones is available. What do I need for this setup? Thanks in advance.Lito ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to mobile phone
btw, i got it, 2N Easygate is highly compatible with Asterisk. Thanks.On 6/27/06, Lito Lampitoc [EMAIL PROTECTED] wrote:what brand of gsm gateway do you think works well with asterisk? On 6/27/06, Colin Anderson [EMAIL PROTECTED] wrote: A GSM gateway will allow you to specify a ruleset so a channel on the gateway is always locked to a particular mobile number, then you just send the call from Asterisk to the gateway and it will do the hunt for you. -Original Message-From: Lito Lampitoc [mailto:[EMAIL PROTECTED]]Sent: Tuesday, June 27, 2006 7:59 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] asterisk to mobile phoneIs it possible to trunk hunt mobile phones in asterisk? say I have one trunkline and 10 mobile phones brought by the engineers in the field, when someone calls the trunkline, asterisk will hunt which of the 10 mobile phones is available. What do I need for this setup? Thanks in advance.Lito ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO for PSTN
If I have 16 PSTN for my trunklines, how many FXO do I need?Thanks.Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO for PSTN
or TDM400P with four FXO modules perhaps?On 28 Jun 2006 01:47:05 -, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 1 FXO per PSTN, so you would need 16 FXO ports.That would be accomplishedby 4 TDM100P with 4 FXO modules on each.Undrhil--- Asterisk UsersMailing List - Non-Commercial Discussion asterisk-users@lists.digium.comwrote:If I have 16 PSTN for my trunklines, how many FXO do I need? Thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Usersmailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO for PSTN
oh sorry, 2 TDM400P with 4 FXO modules each :=)On 6/28/06, Lito Lampitoc [EMAIL PROTECTED] wrote: or TDM400P with four FXO modules perhaps? On 28 Jun 2006 01:47:05 -, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 1 FXO per PSTN, so you would need 16 FXO ports.That would be accomplishedby 4 TDM100P with 4 FXO modules on each.Undrhil--- Asterisk UsersMailing List - Non-Commercial Discussion asterisk-users@lists.digium.comwrote:If I have 16 PSTN for my trunklines, how many FXO do I need? Thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Usersmailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO for PSTN
yes. sorry, wrong computation :=)On 6/28/06, Steven [EMAIL PROTECTED] wrote: No. As far as my maths goes.. 2 TDM400P's with 4 FXO modules each = 8 FXO's = 8 PSTN lines. It's like John said. Very simple maths one would of thought, unless I'm completely off the mark. In which case I do apologise. HTH Steve From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Lito Lampitoc Sent: 28 June 2006 02:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXO for PSTN oh sorry, 2 TDM400P with 4 FXO modules each :=) On 6/28/06, Lito Lampitoc [EMAIL PROTECTED] wrote: or TDM400P with four FXO modules perhaps? On 28 Jun 2006 01:47:05 -, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: 1 FXO per PSTN, so you would need 16 FXO ports.That would be accomplished by 4 TDM100P with 4 FXO modules on each. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: If I have 16 PSTN for my trunklines, how many FXO do I need? Thanks. Lito -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.5/377 - Release Date: 27/06/2006 ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how scalable is digium cards?
This might be a newbie question but I'm just wondering how would it be possible to have 30 analog lines using asterisk for PBX by just using TDM40B and X100P (or are there any device), if an ordinary PC support just 4 PCI slots? the maximum scale i guess would just be 2 x 8. Adding a new PC just for this purpose would be costly. I would appreciate your comments. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk gateway to other gateways
Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for second to enable the dialer to send the whole overseas digit. Assume the caller is not in database, asterik could give user a busy tone, IVR or just leave it and sends out a DTMF A tone anyway. Once the overseas digit are sent from dialer to asterik, asterik will then decide which telco/carrier/Voip to send the traffic to using LCR. Please note that we need to assign at least 5-10 telco/carrier/Voip access number for backup purposes. Once the least cost destination is selected by asterik, asterik will pick up the PRI line and dial a local access number and waits for a DTMF A tone. Once the A tone is heard from telco/carrier/Voip, it will send the overseas digit which was sent by the dialer earlier on. Also, can asterik sends out a musical tone or IVR while connecting to other telco to advice user that the call is connecting, else it would be dead air from there on. The whole process takes less than 5 seconds while the user stays on the line for this whole thing to happen. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
